• Obineg

    the algorithm would be fairly complex. it would not only need to analyze the transfer function (oir its reversed) in various ways, it would also need to know all the filter externals you have installed and how they sound.

    that is basically a deep learning application, which you first have to train and manually correct its output.

    posted in technical issues read more
  • Obineg

    sure, if you know the exact formula which was used to construct the FIR, you can reverse the process.

    not for "plastic cup".

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  • Obineg

    in 32 bit max/msp you could expr for that.

    because expr is not only 64 bit inside and give you a higher precision result from a complex function - it will also save the long form you typed in, since it is characters and not numbers.

    [expr 1.23456789]

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  • Obineg

    clock division and clock multiplication for a phaseaccumulator?

    division is easy.

    [*~ 4.]
    [%~ 1.]

    but multiplication is more or less impossible, as it would require something to count the current subcycle.

    there are a workarounds though. if you want to use a phasor~ and a phasor~ which is 4 times slower, you could

    a ) use multiple phasors~ from the beginning on and then select one of them

    or, if you need them to be in sync with a "master phasor",

    b ) use a phasor which is 8 times slower than the rated rate and derive the base speed phasor from it already using multiplication.

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  • Obineg

    while this (a copy of the OP) works for cross FM, it does not for a feedback loop.

    posted in patch~ read more
  • Obineg

    samplingrate reduction and bitdepth reduction will still not give you satisfying results.

    the AKAI S-900 and S-950 for example have a variable-samplingrate DA converter and analogfilters - these are the main soundforming parts.

    the emulator II still has the SSM filters.

    while you can emulate analog filters quite good today, implementing that AKAI type of DAC within a fixed-rate digital enviroment is not so easy.

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  • Obineg

    i wonder in which context you gonna need this.

    normally it should be fine to move everything which is time critical into the scheduler queue, and move everything out of it which is not so important.

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  • Obineg

    oh pardon, it does not even have any inputs... i would have bet it is the same external as in maxmsp. :/

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  • Obineg

    well, biquad~ of course, if you know how to handle it for different filter types.

    but take care with fast signal rate modulation. :)

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  • Obineg

    this is a really strange approach.

    posted in abstract~ read more
  • Obineg

    and while GUI objects might be able to understand a message to set new min and max values, it is not good practice to use this feature to perform math operations.

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  • Obineg

    you solve this kind of range conversion stuff like you should solve most prgoramming problems anyway: break the problem down into smaller parts.

    1. what range has the GUI element and what does this represent? aha.

    2. what numbers you want to get of it in the end and what do they represent? so so.

    3. now if you dont get the conversion to work by rule of three kind of equations, the chance is high that it is a combination of more than one times rule of three.

    4. and when you notice that "playspeed" has the habit to work differently below 0. than above 0., try to do this conversion using 0.-2. first, and not with the values currently required for a specific app.

    i could tell you something about your specific patch, too, but i dont understand it. (is there no [slide] object for pd?)

    however, in case you are doing somethign with picth modulation it is in most (yet not in all) situations more useful to do what you want to do on the linear layer, i.e. -> ftom -> myalgorithm -> mtof ->

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  • Obineg

    @schitz

    this answer might be a bit terrifying, but i am afraid that what you had to do is to not change anything in the splitted signals.

    (or in other words: only if you do that what happens in an ideal 3-way speaker)

    otherwise any correctly set up bandsplitter - based on a butterworth or FFT - will lead to the usual "noticeable" filter artefacts.

    for things like multiband compression and such you might want to try using phaselinear filters instead. it is not like it would be required, but it is an alternative.

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  • Obineg

    @manuels

    pardon, i was picking out the special case of waveset-releated applications when i was talking about wavetable synthesis, because that is where you often want to calcualte "FM" "PM "AM" "RM" kind of stuff onto existing waves.

    while i might have lost a bit what the original problem was about, i strongly recommend to work with full waves only, because it somehow looks better. for calculation or for visual control a half wave often says nothing about what happens and if your code was right. :)

    tell me on which of the files above i should look again and what problem you want to solve and i´ll try to give a better answer.

    posted in patch~ read more
  • Obineg

    @bocanegra
    [quote]Now try feeding [pipe]'s output back into it's input[/quote]

    sure, but that is not releated to the flush command - IMHO this is not a bug of the pipe object.
    the flush command cant "close the input" before it sends the buffered content out the outlet, or the order at the two inputs would be not ideal.
    and if you do, you get a stack overflow because pipe outputs at scheduler.

    connecting objects to themselves is a special case, and i am not suprised it does not work with any form of "delay 0."

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  • Obineg

    flush works for me as expected, the error must be elsewhere in the patch.

    grafik.png

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  • Obineg

    @bocanegra
    if the patching is causing a stack overflow (havent tried it either) you could in both cases solve that using a gate.

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  • Obineg

    wavesets are often only created unipolar/halfwave style - like in the PPG wave to blofeld line of synthsizers - but today we have enough RAM to do a full wave, and in a language like pd it can become complicated to read out halfwaves as full waves, when the phase accumulator itself should be modulated (doing realtime FM and PM on a wavetable generator and things like that, but also situations of interpolated readout, resetting the phase and so on)

    posted in patch~ read more
  • Obineg

    i believe what you want to do is set pipe´s delay time to 0 and then back to another value. this also should emtpy its buffer.

    besides that, your "web" with the transition states would be better placed in an object like [text] or [coll]. including the random value, that is.

    posted in technical issues read more
  • Obineg

    inversion per pole is not very interesting, but it is also not very difficult: * -1, + 1 if it is >=0, , * -1, - 1 if it is <0.

    for wavesets try: lowpass, highpass, phase distortion (if phase is not available, create one from the length), FM, AM, and RM using a cosine - or shorten it and leave gaps in the unused space.

    then of course crossmix against a completely different wave, layer with other waves, and combiniation of all these basic techniques.

    posted in technical issues read more
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